The delivery of Voice over Internet Protocol (VoIP) is an emerging technology that offers several advantages over the traditional dedicated circuit-switched connections of the public switched telephone network (PSTN). Exemplary advantages that VoIP offers over PSTN include bandwidth consolidation and speech compression, both of which can contribute to overall network efficiency, among other advantages.
However, traditional VoIP solutions employ a “stacked header” approach, such as in the schematic representation of a VoIP frame 100 shown in FIG. 1. VoIP frame 100 includes an Internet protocol (IP) header 110, a user datagram protocol (UDP) header 111, a real time transport protocol (RTP) header 112, and a payload field 120 that carries voice data, such as adaptive multi-rate (AMR) voice data.
Consequently, when transmitting compressed speech, the accumulated size of the protocol headers 110-112 can result in high overhead and inefficient link usage. For example, assuming Adaptive Multi-Rate (AMR) at 7.95 kbps and 35% silence, the resulting VoIP stream bandwidth would be approximately 17.5 kbps, yielding a 3.7:1 reduction (compressed from a 64 kbps stream), while protocol headers would account for over 60% of the bandwidth. Similar examples can yield reduction far less than 4:1 compression, yet many applications require compression of 6:1, 8:1 or greater.